Networking architectures have grown increasingly complex in communications environments. In addition, the augmentation of clients or end users wishing to communicate in a network environment has caused many networking configurations and systems to respond by adding elements to accommodate the increase in networking traffic. As the subscriber base of end users increases, proper routing and efficient management of communication sessions and data flows become even more critical.
Service providers can offer real-time media services [such as voice and video] over Internet Protocol (IP)-based networks. Real-time media services are susceptible to a wide range of quality issues, including packet loss, packet delay, and packet jitter. The Real-time Transfer Protocol (RTP), which is typically used to transport voice and video over IP, and its companion RTP Control Protocol (RTCP) are designed to facilitate computation of quality metrics by the endpoints involved in an RTP session. The basic RTP/RTCP specification offers guidance on how to calculate and communicate packet loss, delay, and jitter information among the RTP session participants. Other specifications offer information on how to calculate and communicate other statistics such as extended reports including voice over IP (VoIP) metrics, audio metrics, and video metrics.